Introduction to Asterisk

1
Introduction to Asterisk

• Or How to spend 2 months on the phone
• John Todd (jtodd_at_loligo.com)
• CTO, VOIP Inc. http//www.voipincorporated.com/
• 2004-09-22 AsterCON, Atlanta GA USA

2
Agenda

• What is Asterisk?
• What is Asterisk NOT?
• What do you want to do? (goals, budget, user
  requirements)
• PBX Replacement
• Super-Brief Examples

3
What is Asterisk?

• a conversion gateway for...
• physical media (C-T1, PRI, FXO, FSX, IP)
• protocol (TDM,SIP,H.323,IAX,MGCP,SCCP)
• codec (G.729,G.711,GSM,ILBC,G.726, etc.)
• an IVR/user interface application server
• a lot more (conferencing, recording, etc.)

4
What is Asterisk? (contd)

• open-source (GPL exceptions)
• blessed (cursed?) with an extremely active user
  community
• easily extended with Perl/C/Python/etc. or apps
  written (typically C)
• flexible enough to do almost any
  telecommunications task (blessing/curse again)

5
What is Asterisk not?

• not a SIP proxy (subtle, yet important)
• not a billing system
• not an OSS (Operational Support System)
• not a natively database-driven system
• not an email tool or USENET browser (yet)
• not easily configured without command-line
  interaction

6
PBX Replacement!

• Primary stated goal is to be a NIX based PBX
  replacement
• Multiple desksets, multiple inbound line
  support (hundreds or thousands)
• Features are comparable to or better than most
  PBX systems (even VoIP-enabled ones) some
  assembly required

7
What do you need to run Asterisk?

• Ugly answer That depends.
• Easy answer Dedicated P4 2.0ghz with good IRQ
  support and 1 X100P card (from Digium at around
  110)
• Linux (RH 9.0, Debian are good choices BSD
  support is there, but shaky)
• Low-jitter, low-loss bandwidth to SIP endpoints
  (desktops and/or upstreams)

8
How big?

• MORE ugly answers That depends.
• If the server is just a SIP redirector, then you
  can scale quite large (tens of thousands?)
• Figure 81 to 101 ratio for offhook users
• Word of the day Erlangs
• Rule of thumb for g.729 transcoding
  2x Xeon 3ghz 100 users

9
Typical VoIP Installation Cost Points

• Server for Asterisk (plus backup, if youre sane)
  - ???
• T1 PRI card for Asterisk (500)
• SIP devices for desktop users (ranges widely -
  figure 120 per user to be safe, for analog
  lines)
• Termination agreement with carrier(s) - ranges
  widely - figure .025 for US traffic, worst-case
  (prices drop radically with volume)

10
CPE

• Analog adapters (VOIP Inc., Sipura, Cisco,
  Grandstream, etc.)
• Typically between 80 and 120 (2 port)
• Digital Handsets (Cisco, Polycom, Snom, Pingtel,
  Grandstream)
• Typically around 300 (YGWYPF)

11
Why are you changing, anyway?

• Implement based on price, expand based on
  features.
• Long Distance will soon become a commodity (i.e.
  invisible) but features of the system will always
  be visible to users
• Integration of telephony into other business
  systems is gradual and subtle start with
  something that is open so you can expand as you
  need.

12
What new stuff are you providing?

• FEATURES! Dont get hung up on building just a
  replacement service. Implement phone
  services which are easily implemented with
  Asterisk (given time, patience, and Perl)
• Sample of services phone spam blocking, inbound
  call redirection based on CLID, time-of-day
  routing, IM integration of VM notices,
  VM-to-email, busy line redirection, multi-number
  custom ringers

13
What do they see?

• Remember the visibility of the customer is very
  limited. They see
• Deskset (equipment) and features
• Call Quality/Call completion
• Price (if theyre the CFO)

14
Non-PBX Use

• Extremely low bandwidth call relay (PRI-to-PRI
  via VoIP) via 802.11b or long-haul WAN
• Dating services/voicemail services
• Text-to-speech service (Nagios, weather, etc.)
• Call centers (inbound or outbound)
• Calling cards

15
Startup Notesor how to really annoy your
spouse/co-workers

• Recommended setup for beginners
• PIII 700mhz or faster machine
• X100P card (Digium 110)
• 2 SIP devices (Sipura, Cisco ATA-186, Cisco
  7960, 40, 05, 12) - 100-300
• Test on your own line or home first, then expose
  to the office

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How it goes together
Channels
SIP
(etc.)
Zap
Context from-sip Extension 1234 Priority 1
Context from-zap Extension (none) Priority 1
Context from-blah Extension 8989 Priority 1
(to extensions.conf)
17
sip.conf
2000 typefriend hostdynamic contextfrom-sip s
ecretmysecret 2001 typefriend hostdynamic co
ntextfrom-sip secretmoresecret
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(calls from SIP channel configs end up here)
extensions.conf
This is where we handle our SIP
calls from-sip exten gt 1234,1,Answer exten gt
1234,2,Playback(tt-monkeys) exten gt
1234,3,Hangup exten gt _20XX,1,Dial(SIP/EXTEN
,30,r) exten gt _20XX,2,Goto(from-sip,EXTEN,102
) exten gt _20XX,102,Voicemail(bEXTEN) exten
gt _20XX,103,Hangup exten gt t,1,Hangup exten
gt h,1,Hangup
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Most-Used Applications

• Dial - tries to make a new call, and then
  connects current channel with new call if
  successful
• Goto - allows arbitrary leaps between contexts
  and priorities allows modification of current
  extension
• Background - plays a file to current channel
  interprets DTMF input

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Magic with Include

• Contexts are NOT parsed in the order they appear
• Break up large contexts into smaller contexts and
  then use include gt ltcontextgt in the main
  context
• This helps your sanity, as well.

21
Wrong
main exten gt _X11,1,Dial(Zap/1/EXTEN,500,r)
exten gt _9.,1,Dial(SIP/EXTEN_at_mysipprovider,60,
r) exten gt _011.,1,Dial(SIP/EXTEN3_at_int-sip,60
,r) exten gt h,1,Hangup
22
Right
main include gt emergency include gt
outside-line include gt international exten gt
h,1,Hangup emergency exten gt
_X11,1,Dial(Zap/1/EXTEN,500,r) outside-line
exten gt _9.,1,Dial(SIP/EXTEN_at_mysipprovider,60,
r) international exten gt _011.,1,Dial(SIP/EX
TEN3_at_int-sip,60,r)
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Links

• http//www.asterisk.org/
• http//www.voip-info/wiki-Asterisk
• http//www.loligo.com/asterisk/
• http//www.onlamp.com/pub/a/onlamp/2003/07/03/aste
  risk.html
• http//www.digium.com/
• http//www.asteriskdocs.org/

24
Unabashed Plug Slide

• VOIP, Inc.
• Builds/Sells MTA SIP hardware (2 port FXS) and
  various other devices
• Sells/Integrates SIP proxy, billing/invoicing
  system, LCR system, customer care system, etc.
  (yes, asterisk is a part)
• http//www.voipincorporated.com/